PSU VoIP blog reader Oskar contributed an updated patch for GTalk shared status/invisible in Asterisk 11. You may recall that I hacked this functionality in to Asterisk 1.8. Oskar’s updates work the same way, now part of res_xmpp instead of the deprecated res_jabber. Thanks, Oskar!

Last week I had the privilege once again of attending Astricon, this time in Atlanta. It was the biggest Astricon yet, showing that Asterisk is continuing to gain users and momentum. As a finale to the show, Digium released Asterisk 11, the latest long-term support (LTS) version. Asterisk 11 includes WebRTC support, ICE/STUN/TURN for NAT traversal, new encryption methods and a reworked Jingle/Google Talk/Google Voice driver set (now called chan_motif).

Digium was heavily promoting their IP phone hardware, giving away D40 sets as quickly as other vendors at the show gave away T-shirts and pens. The phones are solid and feature-full, akin to current models of Polycom and Cisco-Linksys. When integrated with Asterisk or Switchvox, provisioning is incredibly simple, taking advantage of mDNS (Bonjour) for discovery and configuration.

Speaking of Switchvox, I got my first real look at the user interface and slick API that are part of Digium’s SMB turn-key solution. If I just wanted to plunk down some money and get a sophisticated, polished and powerful PBX for a small or medium business, I’d buy it.

I attended a session on controlling Asterisk with HTML5 and Node.js, and another on the Adhearsion platform for controlling Asterisk. Both sessions were inspiring, but I know neither Javascript (for HTML5/Node programming) nor Ruby (for Adhearsion). The only question in my mind is which of these I should learn first.

Get to know your work habits and soon you’ll know that time of day that is your productivity sweet spot.

Mine is 10:00a.m.

Coincidentally, the majority of meetings to which I am invited happen right around this time of day. Unfortunately, meetings aren’t typically where work gets done. They are where work gets talked about. The actual work gets done outside the meetings.

Some time ago I explored FreeSWITCH’s mod_skypopen. The module works great and cleanly connects FreeSWITCH to a normal Skype account. In that posting, I noted that there’s a simple script you can run to set it all up. But the script only works if you are using a mostly default FreeSWITCH install with all the configs in place. If, on the other hand, you have a minimal FreeSWITCH install as a Google Voice gateway to Asterisk, like I do, the script won’t work.

Below is the step-by-step method to configuring mod_skypopen on FreeSWITCH if you’re using it only as a gateway/connector. I had thought that no one would want this and had forgotten about it, until PSU VoIP reader Ranga asked about it. Major thanks to Ranga for testing out the following procedure and providing helpful feedback! I admit, it’s still pretty laborious–not an easy how-to by any definition. But it works as a reliable method of connecting Skype to your Asterisk server.

The procedure: Start here

Log in to your system’s command line interface and perform all the steps as root unless otherwise instructed.

If any step results in an error where the command is not found, you may need to install the program using yum (yum install program name) or build the program from sources on the web. You can find Git at http://git-scm.com/.

Start and login to Skype as regular user through X. This may be tricky. I use SSH tunneling and on my workstation (Mac) I have X11 available to display X programs. If you want to do it at your server console you’ll need to install X11.org or maybe just use vncserver. There are a few options here. Setting up a way to run X programs = exercise left for the reader.

Log out of Skype after successfully logging in once. The Skype config is in /home/username/.Skype .

Copy it to /root (so that you have /root/.Skype). If you just ran Skype as root, the config is already in the right place.

Edit the /root/.Skype/SKYPEUSERNAME/config.xml file and add the bolded lines in before the Reminders tag at the bottom:

Now you have FreeSWITCH connected to Skype and Skype is logged in. It is time to configure Asterisk.

Incoming Skype calls will ring sip:75973@127.0.0.1, so asterisk needs to be listening on 127.0.0.1 (it does by default) and have a route for number 75973. If you write your own Asterisk config files, add some dialplan in extensions.conf to route 75973 to wherever you want. For FreePBX, set up an Inbound Route for DID 75973 and route it where you’d like your incoming Skype calls to go.

Today’s final session of Astricon was an update on Asterisk 10, Asterisk SCF and DAHDI, the Digium hardware driver.

SCF, the scalable distributed framework, will go beta in November or December. The Digium folks pointed the audience to the Asterisk project wiki for more information, so I will do the same and point you to it: Asterisk SCF Home (wiki). SCF was announced at last year’s Astricon and my take on it is that it will make Asterisk enterprise- and carrier-grade.

Asterisk 10 has some great improvements related to media handling, conferencing, and FAX. Digium’s Steve Sokol gave a great interview to TMC blogger Tom Keating: Asterisk 10 – The Inside Story.

Astricon has been a great mix of training, product information, case studies, and networking. Some of the sessions, such as one on how to program using the Asterisk Manager Interface, are clearly Asterisk-specific. Others have dealt with more general topics such as load-balancing VoIP services using a SIP proxy like Open{SER|SIPS}|Kamailio and tying in some cloud services, such as those from Voxeo (I’m a fan), to add voice features or logic. Like last year, the Astricon presentations are being recorded and will be accessible on astricon.net.

Denver is a nice city but has some unusual weather:

Before: Monday afternoon, high 70s, beautiful fall colors

After: Wednesday morning, about 30 degrees, several inches of snow!

I’m all about wikis, forums, and blogs when it comes to learning about Asterisk, but I’m excited to have the official paperback books on the subject:

I have some posts planned, including notes on a manual Skype-FreeSWITCH integration. Next week, I might have a bit to share from Astricon. This will be my first time attending the conference. To my fellow open-source VoIP enthusiasts: see some of you there!

This is the flip side of the coin–the other half of IP telephony and the systems administration dilemma. It’s for the guys like me who enter the arena knowing how to configure, operate and administer a server, with some server-level operating system on it, and who are handy with the tools of that OS. We are adept at building open-source software from the C source files and issuing commands to run the software. It’s for those of us who arrogantly look at software like Asterisk or FreeSWITCH and think, “Piece of cake. Install the build tools, configure, make, edit a config file and done! I’ve done it a million times!”

Not so fast. What the systems administrator has before him or her at that point is software that may run, but not do anything useful. And to make it work in a sensible way, it seems like you have to understand

the whole realm of telephony hardware, including endpoints and interfaces to analog and digital telephone networks;

networking protocols, including some fairly complicated quality-of-service configurations and NAT/firewall workarounds if you’re in such an environment;

phone company lingo, so that you can order the right service from a provider to connect you to the outside world;

technical voice services areas like call routing and dial plans, so that your phone system is actually usable (by your business or family);

business functions, so that you can get the call records into the hands of the people who are going to charge the users for their usage.

And, because I am sure I am forgetting a number of other areas, I’ll tack on “and much more.”

I personally have found these areas much harder to learn than server and operating systems administration. When it comes to voice, I know what I know, and more importantly, I know what I don’t know, which is quite a lot! But the real stumper is when I don’t know what I don’t know. (You know?) In situations like that, Google doesn’t help much because I don’t have the right terminology in the search. It’s time to find the voice gurus out there, the ones who have been through the various iterations of PBXes.

I suspect that it’s easier to add server and OS administration to a full and rich voice services skill set than it is to take a systems administrator and make him/her into a competent voice person. Any commenters care to share opinions on this?